[OpenSIPS-Users] Broken Loose Routing?

Brett Nemeroff brett at nemeroff.com
Thu Oct 30 16:16:41 CET 2008


Ok, that makes a little more sense to me. However, OpenSIPs isn't relaying
it to the public IP which is in the RURI of the BYE, it's sending it to the
private IP on OpenSIPs. Is that because of the Route Header?

So this seriously looks like a proivder problem not adhering to standards?
Is there anythign I can do on my side to make it play nice?
 I'm not sure I can get this provider to change anything .

-Brett

---------- Forwarded message ----------
From: Iñaki Baz Castillo <ibc at aliax.net>
Date: Thu, Oct 30, 2008 at 10:06 AM
Subject: Re: [OpenSIPS-Users] Broken Loose Routing?
To:
Cc: users <users at lists.opensips.org>


2008/10/30 Brett Nemeroff <brett at nemeroff.com>:


> Please note, the provider isn't sending the BYE to the internal address.
The
> provider sends it to OpenSIPs Public address. Which is correct..

No, that's not correct, the RURI of the BYE from provider *MUST* be
the same as the RURI in the INVITE Contact the provider received.
The BYE is sent back to the public IP, of course, but it's routed
thanks to "Route" header (loose route).

This is the INVITE OpenSIPS sends to the provider:


U 62.175.182.141:5060 -> 195.62.163.131:5060
INVITE sip:17475551212 at 195.62.163.131:5060;transport=udp SIP/2.0.
Record-Route: <sip:62.175.182.141
;r2=on;lr;ftag=1c2145666146;did=fe6.729dae35>.
Record-Route: <sip:10.10.65.141
;r2=on;lr;ftag=1c2145666146;did=fe6.729dae35>.
Contact: <sip:15125551234 at 10.10.67.190:5060>.


And this is the BYE the provider sends to OpenSIPS:

U 195.62.163.131:5060 -> 62.175.182.141:5060
BYE sip:15125551234 at 62.175.182.141:5060 SIP/2.0.
Record-Route: <sip:195.62.163.131;lr;ftag=BDA2FF96-481D6819>.
Route: <sip:62.175.182.141;r2=on;lr;ftag=1c2145666146;did=fe6.729dae35>,
<sip:10.10.65.141;r2=on;lr;ftag=1c2145666146;did=fe6.729dae35>.
Contact: <sip:17475551212 at 76.235.229.126 <sip%3A17475551212 at 76.235.229.126>
>.


As you can see:
1) The BYE is sent to the the address in top Route (loose route) -> OK
2) The BYE RURI address is 62.175.182.141:5060 while INVITE Contact is
10.10.67.190 -> ERROR
3) I really wonder why the BYE contains a useless Record-Route.

The behaviour in OpenSIPS is correct. The provider creates a wrong
in-dialog request (BYE) since it should be:
 BYE sip:15125551234 at 10.10.67.190:5060 SIP/2.0


--
Iñaki Baz Castillo
<ibc at aliax.net>
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